mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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${ noResults }
24709 Commits (55e15a2cc92b9c90dbe0d97d20e5a89430016deb)
| Author | SHA1 | Message | Date |
|---|---|---|---|
David M. Lee
|
55e15a2cc9 |
My doc correction in r400842 had a silly bug.
Because I added a wiki_description to models and not their properties, the rendered wiki page had the model description instead of the property descriptions, which looks very silly indeed. (closes issue ASTERISK-22705) ........ Merged revisions 400958 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400959 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
Richard Mudgett
|
2127848d6c |
chan_dahdi: Add config support for hwgain settings.
* Add hwtxgain and hwrxgain config options to chan_dahdi.conf with
documentation in chan_dahdi.conf.sample.
(closes issue ASTERISK-22429)
Reported by: Jaco Kroon
Patches:
jira_asterisk_22429_hwgain_trunk.patch (license #5621) patch uploaded by rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400950 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
12 years ago |
|
Richard Mudgett
|
edb437ee6c |
chan_dahdi: Reflect the set software gain in the CLI "dahdi show channel" output.
* Remember the swgain setting from CLI "dahdi set swgain" command so the
CLI "dahdi show channel" output will reflect the current setting.
* Updated CLI "dahdi set hwgain" and "dahdi set swgain" documentation.
(issue ASTERISK-22429)
Reported by: Jaco Kroon
Patches:
jira_asterisk_22429_v1.8_v2.patch (license #5621) patch uploaded by rmudgett
........
Merged revisions 400907 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........
Merged revisions 400909 from http://svn.asterisk.org/svn/asterisk/branches/11
........
Merged revisions 400911 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
|
12 years ago |
Mark Michelson
|
47e910bfe6 |
chan_sip: Do not increment the SDP version between 183 and 200 responses.
Bumping the SDP version number can cause interoperability problems since receivers of the responses will expect that a 200 SDP will be identical to a previous 183 SDP. (closes issue ASTERISK-21204) reported by NITESH BANSAL Patches: dont-increment-session-version-in-2xx-after-183.patch uploaded by NITESH BANSAL (License #6418) ........ Merged revisions 400906 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400908 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400910 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400912 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
Kevin Harwell
|
61b60fab2d |
pjsip outbound registration: Log message says received a 408 when we didn't
If the server didn't exist that we are trying to register to the log message would say that a 408 was received from that server when in reality one wasn't. Added log messages stating no response was received if the response does not exist. (closes issue ASTERISK-22554) Reported by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2893/ ........ Merged revisions 400890 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400891 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
Matthew Jordan
|
fd4919e466 |
Remove duplicate module info block
The module info block was repeated twice. Once is sufficient. ........ Merged revisions 400881 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400882 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
Joshua Colp
|
b47851264e |
Fix a race condition in res_pjsip_session with rapidly terminating the session.
The INVITE session state callback wrongly assumes that a session will always exist, but when rapidly terminating the session this assumption goes out the window. As all handler code for the INVITE session state callback requires the session it will now just exit immediately if no session exists. (closes issue ASTERISK-22668) Reported by: John Bigelow ........ Merged revisions 400872 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400873 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
Kinsey Moore
|
1a0a2b3e4c |
Fix realm comparison for outbound auth
When generating the list of authentication credentials to pass to PJSIP, Asterisk was using the raw pointer of a pj_str_t which is not always NULL-terminated. This sometimes resulted in incorrect text for the realm and a failure to match the realm for authentication purposes which was causing the outbound nominal auth pjsip basic call test to bounce. This now uses the pj_str_t that contains the realm instead of generating a new one. Thanks to John Bigelow for helping to narrow this down. ........ Merged revisions 400863 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400864 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Richard Mudgett
|
224f44c971 |
channel.h: whitespace changes.
........ Merged revisions 400854 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400855 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
David M. Lee
|
9234804a3b |
Multiple revisions 400508,400842-400843,400848
........
r400508 | dlee | 2013-10-03 23:54:51 -0500 (Thu, 03 Oct 2013) | 1 line
Corrected response class for stopPlayback
........
r400842 | dlee | 2013-10-10 14:23:24 -0500 (Thu, 10 Oct 2013) | 1 line
Correct some ARI wiki rendering errors
........
r400843 | dlee | 2013-10-10 14:26:19 -0500 (Thu, 10 Oct 2013) | 1 line
Updated /play resource docs. The playback of http: resources isn't implemented... yet
........
r400848 | dlee | 2013-10-11 11:18:46 -0500 (Fri, 11 Oct 2013) | 5 lines
Fix a stupid copy/paste error in ARI docs.
Patches:
ari-doc-patch.txt uploaded by jbigelow (license 5091)
........
Merged revisions 400508,400842-400843,400848 from http://svn.asterisk.org/svn/asterisk/branches/12
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12 years ago |
|
David M. Lee
|
07e2b1d028 |
Fixed merge tracking for r400360, which was somehow lost
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400851 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Richard Mudgett
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0ddcee5a46 |
Softmix: Fix crash when switching from softmix to another bridge technology.
The crash is caused by a race condition when switching between native RTP
and softmix bridging technologies. In this situation, the bridging
technology is switched from native RTP to softmix, and then back to native
RTP fast enough that the softmix private data gets destroyed before the
softmix mixing thread gets started.
Thanks to Kinsey Moore for the crash analysis.
* Fix race condition when starting the softmix mixing thread and switching
to another bridge technology.
(closes issue ASTERISK-22678)
Reported by: John Bigelow
Patches:
jira_asterisk_22678_v12.patch (license #5621) patch uploaded by rmudgett
Tested by: John Bigelow
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Merged revisions 400849 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
|
Joshua Colp
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113c281782 |
Perform validation of permanent contacts on AORs in res_pjsip.
........ Merged revisions 400833 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400834 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Joshua Colp
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cbbcf1808c |
Fix an assertion in res_pjsip when specifying an invalid outbound proxy.
This change fixes two issues when setting an outbound proxy: 1. The outbound proxy URI was not parsed and validated during configuration. 2. If an outgoing dialog was created and the outbound proxy could not be set an assertion would occur because the usage count on the dialog was not decremented. The documentation has also been updated to specify that a full URI must be specified for the outbound proxy. (closes issue ASTERISK-22672) Reported by: Antti Yrjola ........ Merged revisions 400824 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400825 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Matthew Jordan
|
fcf6c84666 |
Use 'z' as the format specifier for size_t
Using 'lu' will produce a compiler warning for some versions of gcc and on some architectures. 'z' should be portable as a format specifier for size_t. ........ Merged revisions 400812 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400813 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Matthew Jordan
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b36ef0b412 |
Add PJSIP_HEADER function for manipulation of SIP headers in the PJSIP stack
This patch adds support to the PJSIP stack in Asterisk for SIP header manipulation. Note that this is analagous to SIPAddHeader/SIPRemoveHeader. For PJSIP_HEADER, an incoming supplemental session callback is registered that takes the pjsip_hdrs from the incoming session and stores them in a linked list in the session datastore. Calls to PJSIP_HEADER traverse over the list and return the nth matching header where 'n' is the 'number' argument to the function. When adding a header, the first call creates a datastore and linked list and adds the datastore to the session. The header is then created as a pjsip_hdr and added to the list. An outgoing supplemental session callback then traverses the list and adds the headers to the outgoing pjsip_msg. When removing a header, the list created with PJSIP_HEADER(add,...) is traversed and all matching entries are removed. (closes issue ASTERISK-22498) Reported by: George Joseph patch: res_pjsip_header_funcs_v1.patch uploaded by george.joseph (License 6322) ........ Merged revisions 400771 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400772 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Kinsey Moore
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8021958099 |
Add warning when compiling with iODBC support
When running configure, libiodbc2 development headers will fulfill the
requirement for ODBC development headers, but will not function
properly. This adds a warning when libiodbc2 development headers are
detected instead of unixodbc development headers.
(closes issue ASTERISK-22459)
Reported by: Patrick Maille
Tested by: Walter Doekes
Patches:
issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes (License 5674)
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Merged revisions 400767 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 400768 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 400769 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
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Richard Mudgett
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478c88991e |
app_agent_pool: Fix AMI/CLI AgentLogoff soft preventing agents from logging back in.
* Clear the deferred_logoff flag when an agent logs in. (closes issue ASTERISK-22669) Reported by: John Bigelow ........ Merged revisions 400754 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400759 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Mark Michelson
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2904a198d5 |
Switch from using pjsip_strerror to pj_strerror.
pjsip_strerror is only aware of PJSIP-specific error codes. pj_strerror() is aware of all PJProject error codes and OS-specific error codes. This specifically fixes an oft-seen error in transport configuration code where EADDRINUSE would result in "Unknown PJSIP error 120098" instead of a useful message. ........ Merged revisions 400749 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400750 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Richard Mudgett
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f87086b374 |
app_confbridge: Can now set the language used for announcements to the conference.
ConfBridge now has the ability to set the language of announcements to the
conference. The language can be set on a bridge profile in
confbridge.conf or by the dialplan function
CONFBRIDGE(bridge,language)=en.
(closes issue ASTERISK-19983)
Reported by: Jonathan White
Patches:
M19983_rev2.diff (license #5138) patch uploaded by junky (modified)
Tested by: rmudgett
........
Merged revisions 400741 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 400742 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
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Richard Mudgett
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665ef4c654 |
app_confbridge: Fix duplicate default_user profile.
* Fixed looking in the wrong profiles container to see if the default_user profile is already created in verify_default_profiles(). The bridge profile container is never going to hold user profiles. :) ........ Merged revisions 400723 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400724 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400728 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Kinsey Moore
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4e72ad1b4f |
Fix func_config list entry allocation
The AST_CONFIG dialplan function defined in func_config.c allocates its config file list entries using ast_malloc. List entry allocations destined for use with Asterisk's linked list API must be ast_calloc()d or otherwise initialized so that list pointers are set to NULL. These uses of ast_malloc have been replaced by ast_calloc to prevent dereferencing of uninitialized pointer values when traversing the list. (closes issue ASTERISK-22483) Reported by: Brian Scott ........ Merged revisions 400694 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400697 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400701 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400704 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Kinsey Moore
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4873c11f64 |
Fix STUN crash when using IPv6 any address
Ensure that when chan_sip binds to the IPv6 any address ([::]), IPv4
candidates are also added.
(closes issue ASTERISK-21917)
Reported by: Torrey Searle
Patches:
0023_ipv6_stun_crash.patch uploaded by Torrey Searle (License 5334)
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Merged revisions 400681 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 400682 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
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Mark Michelson
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a4d6f43962 |
Push CLI qualify into the threadpool.
If you run Asterisk in the background and then connect to it through a separate console, the thread that runs CLI commands is not registered with PJLIB. Thus PJLIB does not like it when you attempt to send OPTIONS requests from that thread. So now we push the task into the threadpool, which we know to be registered with PJLIB. Thanks to Antti Yrjola for reporting this. ........ Merged revisions 400680 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400683 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Richard Mudgett
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8eec8fbf83 |
Make app_queue and res_agi independent of AMI being enabled.
The https://reviewboard.asterisk.org/r/2888/ review changes manager to not subscribe to stasis when it is disabled for performance reasons. When manager is disabled app_queue and res_agi decline to load and fail to clean up what they have already allocated. * Made app_queue and res_agi clean up allocated resources when they decline to load. * Made app_queue and res_agi use their own subscriptions to the stasis topics instead of borrowing manager's message router structure inappropriately. (closes issue ASTERISK-22604) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/2902/ ........ Merged revisions 400671 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400672 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Richard Mudgett
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96d27333d2 |
Miscellaneous stand alone comment cleanups.
........ Merged revisions 400661 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400662 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
Michael L. Young
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0e218f76e2 |
app_queue: Fix Queuelog EXITWITHKEY only logging two of four fields
Commit r62462 added two extra fields for logging "the original position the
caller entered the queue at, and the amount of time the caller was waiting in
the queue." But when r75969 was merged from 1.4 into trunk (r75977), these two
fields disappeared. Those two extra fields were not logged in 1.4 and when the
patch was merged, those fields went away.
Therefore, this is a regression and was caught by the reporter because he was
reading the awesome "Asterisk: The Definitive Guide" book.
(closes issue ASTERISK-22197)
Reported by: Dalius M.
Tested by: Dalius M.
Patches:
asterisk-22197-q-log-exitwithkey.diff
uploaded by Michael L. Young (license 5026)
Review: https://reviewboard.asterisk.org/r/2901/
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Merged revisions 400622 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 400623 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 400624 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
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Richard Mudgett
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e2a0a6ed19 |
chan_iax2: Fix compile error.
........ Merged revisions 400588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400593 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Michael L. Young
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2af53640c8 |
Add IPv6 Support To chan_iax2
This patch adds IPv6 support to chan_iax2. Yay! (closes issue ASTERISK-22025) Patches: iax2-ipv6-v5-reviewboard.diff by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2660/ ........ Merged revisions 400567 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400568 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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David M. Lee
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21902bf964 |
Added missing file from r400522
........ Merged revisions 400552 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400553 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
Jonathan Rose
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f4ebdca52a |
chan_pjsip: Make logger togglable without loading/unloading
This patch makes the res_pjsip_logger do a few things... First, it
will be built and installed by default now, so end users won't need
to enable it in menuselect. Second, while it is loaded, it no longer
will immediately issue log messages. Upon loading, it is in the
disabled state and must be turned on with the new CLI command. The
CLI command 'pjsip set logger <on/off/host> has been added and can be
used to do the following:
pjsip set logger on:
Enables logger for all PJSIP traffic
pjsip set logger off:
Disables logger for all PJSIP traffic
pjsip set logger host <host>:
Enables logger for the specific host
Review: https://reviewboard.asterisk.org/r/2900/
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Merged revisions 400542 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400543 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
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Jonathan Rose
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44bd543181 |
chan_pjsip: Add alembic scripts for generating db tables for PJSIP
Also updates sample configurations for sorcery and extconfig to demonstrate how to use databases created by that alembic script. (closes issue ASTERISK-22133) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2892/ ........ Merged revisions 400532 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400533 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Matthew Jordan
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8d7873b836 |
ARI: Add subscription support
This patch adds an /applications API to ARI, allowing explicit management of
Stasis applications.
* GET /applications - list current applications
* GET /applications/{applicationName} - get details of a specific application
* POST /applications/{applicationName}/subscription - explicitly subscribe to
a channel, bridge or endpoint
* DELETE /applications/{applicationName}/subscription - explicitly unsubscribe
from a channel, bridge or endpoint
Subscriptions work by a reference counting mechanism: if you subscript to an
event source X number of times, you must unsubscribe X number of times to stop
receiveing events for that event source.
Review: https://reviewboard.asterisk.org/r/2862
(issue ASTERISK-22451)
Reported by: Matt Jordan
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Merged revisions 400522 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
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Joshua Colp
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b52c972b17 |
Enclose the To URI and update its user portion if a request user has been specified.
........ Merged revisions 400520 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400521 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Joshua Colp
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47da03e737 |
Replace the connection address at the SDP level if altering the SDP with the external media address.
........ Merged revisions 400510 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400511 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Jonathan Rose
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d35b5c9cb0 |
chan_sip: Don't ignore expires value in contact header if it lacks semicolon
(closes issue ASTERISK-22574)
Reported by: Filip Jenicek
Patches:
chan_sip_expires.patch uploaded by Filip Jenicek (license 6277)
........
Merged revisions 400469 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 400470 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 400471 from http://svn.asterisk.org/svn/asterisk/branches/12
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12 years ago |
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Matthew Jordan
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facebb3f3c |
Remove publication of a channel snapshot when the technology is set
This patch removes said publication for a few reasons: (1) It is unnecessary. Association of the channel technology with a specific channel is an implementation detail that should be assumed to "just happen", and consumers of Stasis don't need to be informed about it. (2) Publication of said message can now cause crashes, as the actual creation of a channel in normal locations now stages its messages. As a result, things that create dummy channels (such as the SIP RTP QOS unit test) and associate them with a channel technology were now crashing, as the channel itself was not known by Stasis. ........ Merged revisions 400460 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400461 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Mark Michelson
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b11983d480 |
Fix assumption in bridge_native_rtp.c regarding number of participants in a bridge.
When a party leaves a bridge, there may be more participants in the bridge than expected. As such, it is important not to make assumptions regarding the list of channels in a bridge. This change makes it so that when a party leaves a native RTP bridge, we unbridge it and the party it was bridged with. Previously, the first and last channels in the list were unbridged since it was assumed that these were the two channels that had been bridged. As previously stated, a new party had been inserted into the bridge, so this logic did not work properly. (closes issue ASTERISK-22615) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2899 ........ Merged revisions 400403 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400452 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Joshua Colp
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9826923805 |
When serializing CDR variables (like for "core show channels") don't output an error if CDRs aren't enabled.
........ Merged revisions 400442 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400443 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Kinsey Moore
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fe4cd68444 |
Fix security events for AMI invalid password
In r337595, additional security events were added for chan_sip authentication failures. The new IEs added to the existing invalid password event were defined as required IEs, but existing users of the event did not set the new IEs and could not since they didn't apply to existing uses. They are now marked as optional IEs. (closes issue ASTERISK-22578) Reported by: Matt Jordan ........ Merged revisions 400421 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400440 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400441 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
|
Joshua Colp
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bd8cf10d60 |
Fix a crash caused by muting and unmuting a channel in ARI without specifying a direction.
(closes issue ASTERISK-22637) Reported by: Scott Griepentrog Patch by Matt Jordan, whose office I have taken over in the name of Canada. ........ Merged revisions 400401 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400402 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Richard Mudgett
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7e698f1f42 |
cel: Some whitespace cleanups
........ Merged revisions 400398 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400399 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Kinsey Moore
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c3b1d8f8c1 |
res_rtp_multicast: Ensure SSRC is set properly
This fixes a bug where the SSRC field on multicast RTP can be stuck at
0 which can cause problems for endpoints trying to make sense of
incoming streams.
(closes issue ASTERISK-22567)
Reported by: Simone Camporeale
Patches:
22567_res_mulitcast_ssrc.patch uploaded by Simone Camporeale (License 6536)
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Merged revisions 400393 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 400394 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 400395 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
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Kinsey Moore
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3578bc30fd |
Detect and use xsltCleanupGlobals when available
This introduces usage of an additional libxslt cleanup function,
xsltCleanupGlobals, when the configure script detects that it is
available. Early versions of the library did not include this function.
(closes issue ASTERISK-22570)
Reported by: Corey Farrell
Patches:
xsltCleanupGlobals.patch uploaded by Corey Farrell (License 5909)
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Merged revisions 400384 from http://svn.asterisk.org/svn/asterisk/branches/12
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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12 years ago |
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Richard Mudgett
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e3d69a5628 |
chan_vpb: Make compile again.
........ Merged revisions 400373 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400374 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Mark Michelson
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f945c6a207 |
Get rid of uses of stasis_topic_wait()
........ Merged revisions 400362 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400364 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Mark Michelson
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ee21eee7e0 |
Cache string values of formats on ast_format_cap() to save processing.
Channel snapshots have string representations of the channel's native formats. Prior to this change, the format strings were re-created on ever channel snapshot creation. Since channel native formats rarely change, this was very wasteful. Now, string representations of formats may optionally be stored on the ast_format_cap for cases where string representations may be requested frequently. When formats are altered, the string cache is marked as invalid. When strings are requested, the cache validity is checked. If the cache is valid, then the cached strings are copied. If the cache is invalid, then the string cache is rebuilt and copied, and the cache is marked as being valid again. Review: https://reviewboard.asterisk.org/r/2879 ........ Merged revisions 400356 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Joshua Colp
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c977f73e13 |
Fix crashes in res_pjsip_sdp_rtp and res_pjsip_t38 when a stream is rejected and external_media_address is set.
The callback function for changing the media address in streams wrongly assumes that a connection line will always be present. This is false as no line is present if a stream has been rejected. (closes issue ASTERISK-22645) Reported by: Rusty Newton ........ Merged revisions 400360 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400361 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Mark Michelson
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addbf276f5 |
Multiple revisions 400318-400319
........ r400318 | mmichelson | 2013-10-02 17:08:49 -0500 (Wed, 02 Oct 2013) | 12 lines Remove unnecessary waits from stasis. Since caches are updated on publisher threads, there is no need to wait for the cache updates to occur after a stasis message is published. In the case of chan_pjsip device state changes, this set of changes caused an improvement to performance. Review: https://reviewboard.asterisk.org/r/2890 ........ r400319 | mmichelson | 2013-10-02 17:10:54 -0500 (Wed, 02 Oct 2013) | 3 lines Remove svn:mergeinfo property. ........ Merged revisions 400318-400319 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400335 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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Michael L. Young
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e4ed9886e6 |
Cast Integer Argument To Unsigned Char
The member reg in the peercnt structure is an unsigned char and peercnt_modify() is expecting an unsigned char argument which gets assigned to peercnt->reg. This patch fixes that by casting the integer argument being passed to peercnt_modify to unsigned char. ........ Merged revisions 400314 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 400315 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 400316 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400317 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |